[..] |
ASTERISK-03000: Latest CVS of app_disa fails to compile |
ASTERISK-03001: [patch] authenticate by IP address only in h323 |
ASTERISK-03002: [patch + bug] Pressing "0" during voicemail review kicks user out. |
ASTERISK-03003: [patch] app_disa does not set callerid properly (new structs) |
ASTERISK-03004: Unknown sip user, crash asterisk res_config_mysql |
ASTERISK-03005: voicemail server send RTP packets before receive ACK |
ASTERISK-03006: issuing reload to add more iax2 peers causes all iax2 activity to die |
ASTERISK-03007: [patch] Make libpri compile on solaris |
ASTERISK-03008: no errno support in cdr_csv.c |
ASTERISK-03009: [patch] Solaris update to support X86 (INTEL) |
ASTERISK-03010: Asterisk segfaults when both res_mysql and res_odbc are configured |
ASTERISK-03011: [patch] Voicemail lets you set your password to "#" |
ASTERISK-03012: [PATCH] don't clone app_groupcount variables in ast_do_masquerade |
ASTERISK-03013: WaitForSilence dies before voicemail stops playing greeting |
ASTERISK-03014: [request] externip DNS query |
ASTERISK-03015: Chan_h323 dials out incorrectly again |
ASTERISK-03016: [patch] Second Newchannel event in manager interface |
ASTERISK-03017: Show dialplan does not support hints |
ASTERISK-03018: "." pattern character only matches first digit with PhoneJACK |
ASTERISK-03019: Speex with Xten phone is buggy (Zap dialed part hears silence) |
ASTERISK-03020: Missing CLID on H.323? |
ASTERISK-03021: Changing of Voicemail Password with asterisk realtime mysql database |
ASTERISK-03022: Core dump with extconfig and sip register and res_config_mysql |
ASTERISK-03023: operator=yes isn't being recognized |
ASTERISK-03024: utils/Makefile installs astman and then overwrites it with smsq |
ASTERISK-03025: [patch] Permit retransmission interval to be configured |
ASTERISK-03026: [patch] Use bit masking with options in sip_pvt, sip_user, and sip_peer |
ASTERISK-03027: acl.c:ast_ouraddrfor is grossly complex and can be significantly simplified |
ASTERISK-03028: [patch] Feature request for turning off display/logging of manager logins |
ASTERISK-03029: [patch] SIP register problems on computers with dynamic IP addresses |
ASTERISK-03030: [patch] Fix BYE for the SIP ack patch |
ASTERISK-03031: [patch] Change type to channeltype (as in chan_sip) |
ASTERISK-03032: [patch] Fix code comments, unregister of apps and changge cli commands |
ASTERISK-03033: [bug + patch] Compile fails with cvs head on mac osx jaguar 10.3 |
ASTERISK-03034: [PATCH] compute the caller's timeout only once |
ASTERISK-03035: [PATCH] convert app_queue to use flag macros |
ASTERISK-03036: [PATCH] store full interface name for members |
ASTERISK-03037: [PATCH] use is_our_turn function in wait_our_turn |
ASTERISK-03038: [PATCH] optimize options parsing in try_calling |
ASTERISK-03039: [patch] Makefile unportable (corrected) |
ASTERISK-03040: [patch] Remove more // comments |
ASTERISK-03041: [PATCH] allow ext_strncpy to return string length |
ASTERISK-03042: [PATCH] config category inheritance |
ASTERISK-03043: [PATCH] make list locking macros consistent |
ASTERISK-03044: [patch] voice mail cuts off on openbsd |
ASTERISK-03045: [patch] make fails on openbsd/sparc64 |
ASTERISK-03046: [Not A Bug] New feature request! |
ASTERISK-03047: sip registrations fail if there are too many regiser => entries |
ASTERISK-03048: don't accept h.323 calls |
ASTERISK-03049: codec preference doesn't work with mysql sipfriends |
ASTERISK-03050: [patch] Support for Greek language (say.c, app_voicemail.c) |
ASTERISK-03051: [PATCH] use new struct ast_flags type |
ASTERISK-03052: CVS stable CVS-v1-0-12/20/04-10:19:13 can't call in after upgrade |
ASTERISK-03053: voicemail accounts with realtime mysql |
ASTERISK-03054: [patch] GSM compilation on PPC64 platform |
ASTERISK-03055: [patch] hold/resume does not work anymore on cisco |
ASTERISK-03056: Queue does not seem to consider members's 'penalty' |
ASTERISK-03057: [patch] No Newcallerid manager events on callerid changes in chan_zap.c |
ASTERISK-03058: [patch] manager events callerid inconsitency |
ASTERISK-03059: [patch] add new/old count to MessageWaiting manager event |
ASTERISK-03060: [patch] Add '*' directory exit option (adds to #2995) |
ASTERISK-03061: [patch] Fix for voice timestamp prediction in IAX2; prevent ts jumps. |
ASTERISK-03062: [patch] cleans up the rest of the "if (audiofd && ctrlfd)" entries in say.c |
ASTERISK-03063: Incorrect pronunciation of certain numbers in Spanish |
ASTERISK-03064: Cisco ATA 18x DTMF Probrem |
ASTERISK-03065: can't play files via IAX if zaptel + T1/E1 board drivers loaded and no span lines defined in zaptel.conf |
ASTERISK-03066: [patch] Use ast_flag macros for chan_iax2 |
ASTERISK-03067: [patch] nat=yes in the general section doesnt work for all users |
ASTERISK-03068: [request] dtmfmode=inband tone duration |
ASTERISK-03069: [PATCH] Sip.conf lists auth as a possible option, but chan_sip.c does not use it. |
ASTERISK-03070: Asterisk cannot say negative numbers |
ASTERISK-03071: Polycom IP300/500 stop ringing when "180 Ringing" is sent |
ASTERISK-03072: [patch] Use ast_flag macros for app_voicemail && fix a buglet |
ASTERISK-03073: [patch] app_curl |
ASTERISK-03074: [patch] Wrong US tone freqs in chan_vpb.c |
ASTERISK-03075: [patch] udev-- an update to README.udev |
ASTERISK-03076: [patch] utilize IE 115 (original called number) |
ASTERISK-03077: [PATCH] modify AST_LIST_REMOVE_HEAD to return removed entry |
ASTERISK-03078: [PATCH] add AST_FLAGS_ALL mask and ensure flags are unsigned |
ASTERISK-03079: sip unable to create/find channel |
ASTERISK-03080: [PATCH] add doxygen docs for list macros |
ASTERISK-03081: [PATCH] make AST_LIST_TRAVERSE safe against entry modification |
ASTERISK-03082: make clean does not remove built bins |
ASTERISK-03083: [PATCH] better alternative to soxmix |
ASTERISK-03084: [patch] SIP failure on multihomed setup (intra&inter), -patch included- |
ASTERISK-03085: New App Curl - Compilation Error |
ASTERISK-03086: Elapsed Time does not increment on bridged calls |
ASTERISK-03087: [PATCH] correct logic error from bug 3130 |
ASTERISK-03088: libs installed in /usr/lib get wrong SELinux context; asterisk will not run |
ASTERISK-03089: Version 1.42 of res_musiconhold ignore music files paths |
ASTERISK-03090: [patch] preliminary TR08 protocol support |
ASTERISK-03091: preliminary TR08 protocol support |
ASTERISK-03092: Upgrading glibc library to 2.3.4 on FC3 causes Asterisk to crash |
ASTERISK-03093: [patch] Adds POSTing ability to new app_curl |
ASTERISK-03094: Sipgate don't work anymore with newest Asterisk |
ASTERISK-03095: [patch] new curl inclusion blows up build with RH 7.3 curl-7.9.5 gcc 3.2.2 |
ASTERISK-03096: [patch] Make app_while compile on FreeBSD |
ASTERISK-03097: [patch] chan_agent formatting, documentation, logging |
ASTERISK-03098: [patch] app_queue formatting, documentation |
ASTERISK-03099: [patch] First digit lost or poorly played in SayDigit, mainly in spanish. |
ASTERISK-03100: Integer overflow in pbx_builtin_wait |
ASTERISK-03101: [patch] Fix debug/verbose error messages at startup... |
ASTERISK-03102: [patch] app_queue musiconhold config option changed |
ASTERISK-03103: Compile fails on apps/app_while.c with GCC-2.95.4 |
ASTERISK-03104: [patch] Set NORMAL_CLEARING cause on calls answered |
ASTERISK-03105: [PATCH] list macros don't need "type" specified |
ASTERISK-03106: [PATCH] change ASTOBJ APIs |
ASTERISK-03107: [patch] Allow the switch statement to contain variable expressions to be evaluated at runtime |
ASTERISK-03108: SIP transfers leave * unresponsive |
ASTERISK-03109: [patch] chanisavailable(): Reformat help text, add missing variable |
ASTERISK-03110: [patch] res_monitor: help text update, ast_safe_system +res_features changes |
ASTERISK-03111: [patch] Immediate option is parsed twice |
ASTERISK-03112: [patch] rtp.c formatting fixes |
ASTERISK-03113: [patch] Speex configuration broken for ABR. |
ASTERISK-03114: Result returning from exec in AGI |
ASTERISK-03115: [patch] Log Execution Stack to Channel Variables |
ASTERISK-03116: information is missing on how to handle Directory() exit event |
ASTERISK-03117: [patch] Fix descriptions in AGI; also alphabetize the help listings. |
ASTERISK-03118: hangup while in AGI and manager Redirect event |
ASTERISK-03119: [patch] AGIs continue even if a file was not found. |
ASTERISK-03120: [patch] Increase the stack size for searching through contexts |
ASTERISK-03121: [PATCH] ASTOBJ API correction |
ASTERISK-03122: Asterisk crashes when SIP client tries to register |
ASTERISK-03123: "Hold" implementation on Asterisk does not comply with www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-07.txt |
ASTERISK-03124: [patch] zconfig.h kernel version check incorrect for 2.4.20 |
ASTERISK-03125: [patch] New app: Show channeltypes |
ASTERISK-03126: [patch] Make sure sin_family is filled in |
ASTERISK-03127: Bug# 0003167 Creates GCC-2.95.4 problem with __builtin_expect |
ASTERISK-03128: [patch] one of 3162's commits introduced a type-o while fixing other similar ones |
ASTERISK-03129: [patch] Summarize conflicts at end of 'make update' |
ASTERISK-03130: [patch] Bug in causecodes |
ASTERISK-03131: [patch] Let SendDTMF specify a timeout |
ASTERISK-03132: [patch] Individual voicemail boxes no longer inherit all options from general section |
ASTERISK-03133: SIP Calls aren't destroyed |
ASTERISK-03134: [PATCH] app_qcall.c fscanf format eror |
ASTERISK-03135: Crash in voicemail |
ASTERISK-03136: DUNDI replies always over 2000ms |
ASTERISK-03137: provide hold music to the calling party until answered, USING 'X' AS FILE |
ASTERISK-03138: __builtin_expect |
ASTERISK-03139: [PATCH] centralize common options, switch more opts to flags |
ASTERISK-03140: [patch] agent logins should persist through an asterisk restart |
ASTERISK-03141: [patch] Enhancements to asterisk.c:set_priority: Get RT without runaway risk? |
ASTERISK-03142: [H.323] Audio portion of Call Failing |
ASTERISK-03143: [patch] Allowto set the Bridge DTMF monitor in app_dial |
ASTERISK-03144: [PATCH] astobj.h will not compile on gcc 2.95 |
ASTERISK-03145: [patch] Prevent possible race |
ASTERISK-03146: [patch] Remove "? update.out" message on "make update" |
ASTERISK-03147: [patch] Inconsistency in documentation of ANSWEREDTIME variable |
ASTERISK-03148: When start asterisk with MOH support in Mac OS X, produce a memory allocation warning |
ASTERISK-03149: [PATCH] fix compile error under GCC 2.95 |
ASTERISK-03150: app_waitforsilence.c and privacy.conf.sample have DOS line endings |
ASTERISK-03151: asterisk/channels/chan_phone.c v1.39 does not compile |
ASTERISK-03152: [patch] Add a prompt of the count of user when joining a meetme |
ASTERISK-03153: Missing Schema/DB dump for res_odbc |
ASTERISK-03154: [patch] sip reload deadlock |
ASTERISK-03155: [patch] fix the horrible, horrible formatting in dsp.c |
ASTERISK-03156: [patch] Record name and playback on enter and leave of conference |
ASTERISK-03157: [Patch] Update res_odbc to correct lack of CLI commands help |
ASTERISK-03158: [PATCH] ASTOBJ_CONTAINER_TRAVERSE doesn't allow early exit |
ASTERISK-03159: [PATCH] support filtering of SIP "show users" and "show peers" |
ASTERISK-03160: [PATCH] support filtering of IAX "show users" and "show peers" |
ASTERISK-03161: [PATCH] filter out unplayable files for moh_files, add CLI command to list files |
ASTERISK-03162: [patch] Voicemail formatting, cli command change |
ASTERISK-03163: [patch] app_test.c debugging, help text |
ASTERISK-03164: SIP registrations fail |
ASTERISK-03165: [patch] Print groups in dumpchan() |
ASTERISK-03166: chan_h323 blocks asterisk start-up |
ASTERISK-03167: [patch] Fix 'show dialplan exten@context' to show extensions in patterns and in included contexts |
ASTERISK-03168: [patch] Add a CLI command to send SIP notify to peers (ex:reboot phone) |
ASTERISK-03169: [request] ability to set non-optimized codec priority |
ASTERISK-03170: incoming call routed via gatekeeper fail |
ASTERISK-03171: [PATCH] support lockless ASTOBJ FIND and FIND_UNLINK |
ASTERISK-03172: [patch] Music on hold is terminated immediately after caller is parked. |
ASTERISK-03173: [patch] Initial support for IAX2 Receiver Reports in chan_iax2.c |
ASTERISK-03174: [PATCH] refactor and clean up code |
ASTERISK-03175: Queue requires an argument |
ASTERISK-03176: [patch] danish/german speech for SayUnixTime |
ASTERISK-03177: app_voicemail needs to be loaded before channels |
ASTERISK-03178: [patch] Attended Pound Transfers 2005 Style |
ASTERISK-03179: [PATCH] change load order for res_musiconhold, add example for moh_files |
ASTERISK-03180: [PATCH] "sip notify", reworked |
ASTERISK-03181: ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
ASTERISK-03182: channel format set wrong on manager redirect |
ASTERISK-03183: Asterisk Core dumps upon reload, Not always |
ASTERISK-03184: [patch] chan_mgcp formatting, spacing cleanups |
ASTERISK-03185: [PATCH] Asterisk crashes when accessing simultanously the mysql module |
ASTERISK-03186: app_dial only hangup after timeout |
ASTERISK-03187: Asterisk SIP/SDP RFC Violation -> One way voice |
ASTERISK-03188: [patch] improve rfc compliance of presence events |
ASTERISK-03189: [patch] Add the ability to pause (temporarily suspend) queue members |
ASTERISK-03190: Many sound files need binary mode in CVS |
ASTERISK-03191: [PATCH] Convert app_dial to use flags - tweaks for chan_iax2 and chan_sip |
ASTERISK-03192: Disposition is being stored as numeral. |
ASTERISK-03193: Problem with quotes in System() |
ASTERISK-03194: Problem with completion of "channeltypes" |
ASTERISK-03195: Call in Queue is hung up after 60 sec |
ASTERISK-03196: dial not respect the t and T options |
ASTERISK-03197: [patch] Make sure we do del any remaining connections + misc fixes |
ASTERISK-03198: [patch] [request] permit softhangup to operate on partial channel name |
ASTERISK-03199: SIP stack stops working when externhost is set and DNS resolution fails |
ASTERISK-03200: [patch] One Touch Record Fix |
ASTERISK-03201: [patch] new api call for app.c |
ASTERISK-03202: [patch] SIP realm based authentication |
ASTERISK-03203: [patch] Typo in MOH error output |
ASTERISK-03204: [patch] res_monitor tweak |
ASTERISK-03205: [patch] Logic bug causes PRI to block on first channel |
ASTERISK-03206: [patch] Make SIP set the BLINDTRANSFER var + "preferred codec" logging change |
ASTERISK-03207: [PATCH] ASTCC Expiration Bounty |
ASTERISK-03208: [patch] res_agi.c Formatting and fastagi |
ASTERISK-03209: [patch] Typo in app_getcpeid.c |
ASTERISK-03210: MGCP does not follow correct call flow when going off-hook |
ASTERISK-03211: [patch] Formatting and logging changes |
ASTERISK-03212: [PATCH] move channel variable inheritance to API |
ASTERISK-03213: [patch] Formatting of source code + cli_unregister |
ASTERISK-03214: [patch] Change for easier MultiLanguage * setup in sounds file |
ASTERISK-03215: [PATCH] utils.c uses snprintf but does not include stdio.h |
ASTERISK-03216: [PATCH] implement new method of setting channel variables from config files |
ASTERISK-03217: [PATCH] make flags macros warn users if they use signed variables |
ASTERISK-03218: [patch] Clean up to use consistent "flagification" |
ASTERISK-03219: Multiple SIP registrations from the same phone fail |
ASTERISK-03220: [request] Timeout for Connect? |
ASTERISK-03221: Speex Makefile header location |
ASTERISK-03222: [patch] Give Zaptel and libpri the same 'make update' functionality as Asterisk |
ASTERISK-03223: why does priindication_oob default to 0? |
ASTERISK-03224: [PATCH] simplify ASTOBJ lifetime mgmt API |
ASTERISK-03225: [Patch] Alert-info -> Alert-Info for broken OptiPoint phones |
ASTERISK-03226: [PATCH] flags conversion, formatting cleanup, minor changes |
ASTERISK-03227: making asterisk-addons crashes |
ASTERISK-03228: [patch] AgentCallbackLogin problem |
ASTERISK-03229: PrivacyMgr will not work on ISDN4Linux |
ASTERISK-03230: Sintax problem on a file. |
ASTERISK-03231: [patch] sip.conf.sample cleanup |
ASTERISK-03232: res_perl won't compile on CVS HEAD |
ASTERISK-03233: [patch] Playtones help text |
ASTERISK-03234: [PATCH] ASTCC Maintainence Fee Bounty |
ASTERISK-03235: chan_iax2.c error while running |
ASTERISK-03236: [patch] variables not set correctly |
ASTERISK-03237: [patch] Debug mode in ztcfg no longer requires re-compile to activate, command line option added (twisted) |
ASTERISK-03238: file_retrieval() failure when using ODBC Storage |
ASTERISK-03239: AGI scripts with bad interpreters reported as "no such file" |
ASTERISK-03240: asterisk send wrong failure code when no supported codec found |
ASTERISK-03241: Registration of Snom 190 failing |
ASTERISK-03242: asterisk fails to start with latest app_queue if you have a queue in queues.conf file |
ASTERISK-03243: [patch] SetVar should not allow readonly variables to be set |
ASTERISK-03244: [patch] README.iax update |
ASTERISK-03245: asterisk deadlock in queueing engine |
ASTERISK-03246: [patch] Race condition in zaptel.c causes a kernel panic & fix |
ASTERISK-03247: [patch] new app RetryDial in app_dial.c |
ASTERISK-03248: Bugfix 3242 stops moh functionality if modules.conf.sample was used |
ASTERISK-03249: Bug in dsp.c with BUSYDETECT_TONEONLY |
ASTERISK-03250: Asterisk Segmentation Fault - layer3.c/mpg123 |
ASTERISK-03251: [patch] Config options and global realm fix |
ASTERISK-03252: [patch] Uninitialized string used in app_privacy |
ASTERISK-03253: VMAuthenticate not working |
ASTERISK-03254: [patch] Get Zaptel modules ready for module_param in Linux 2.6 kernel |
ASTERISK-03255: cannot us IVR on incoming IAX2 LINE |
ASTERISK-03256: [patch] DYNFS is not correctly set |
ASTERISK-03257: Wrong response to MESSAGE to non exisiting user |
ASTERISK-03258: asterisk opens a socket for each incoming MESSAGE and crashes when the limit (eg 500) is reached |
ASTERISK-03259: [patch] realtime intermittent SQL lookup failure |
ASTERISK-03260: SIP phone transfer an IAX caller to another SIP phones causes asterisk to crash. |
ASTERISK-03261: [patch] app_dial Fix |
ASTERISK-03262: [patch] When group= gets error parsing, it returns 0 instead of the result. |
ASTERISK-03263: Asterisk hangs if started with -c and then losing terminal |
ASTERISK-03264: SIP phones cannot register after 01/09/2005 CVS (Realtime) |
ASTERISK-03265: [sounds] New prompts |
ASTERISK-03266: [patch] zaptel no longer compiles under 2.6 |
ASTERISK-03267: [patch] a playlist function for ast_streamfile |
ASTERISK-03268: patch: smsq.c does not use correct queue name when sub address used |
ASTERISK-03269: [patch] Normalization of RFC2833 events is not robust |
ASTERISK-03270: [patch] Fix and simplify loguniqueid, fix reload, avoid deadlock on malloc failures |
ASTERISK-03271: [patch] Centralize all iax2 peer lookups to the find_peer() function |
ASTERISK-03272: [patch] Update unused outdated debug code |
ASTERISK-03273: [patch] Zaptel doesn't compile under 2.4 kernel |
ASTERISK-03274: [patch] clone_variables crashes on masq of channel with no vars |
ASTERISK-03275: [patch] fxotune.c doesn't check wheter to include <linux/zaptel.h> or "zaptel.h" |
ASTERISK-03276: [patch] PREFERRED_CODEC - codec preference override |
ASTERISK-03277: [patch] convertion //comments ---> /* comments */ to jive w/ coding-guidlines |
ASTERISK-03278: Polycom Phones Unable To Complete Calls |
ASTERISK-03279: Multiple hours/days/weeks for GotoIfTime |
ASTERISK-03280: [Patch] Loss of callerid when calling from chan_vpb.c |
ASTERISK-03281: [patch] increase group number limitation |
ASTERISK-03282: [PATCH] support completion of peer names for "sip show peer" and "sip debug peer" |
ASTERISK-03283: h323 channel compile broken PPC |
ASTERISK-03284: zaptel no longer compiles under SuSE kernel 2.6.5 |
ASTERISK-03285: [PATCH] new ASTOBJ implementation, including hash-table containers |
ASTERISK-03286: [PATCH] optimize module loading |
ASTERISK-03287: [patch] function to handle more easily, argument for applications |
ASTERISK-03288: [patch] Force a make clean after cvs updates |
ASTERISK-03289: [patch] astDB convenience settings for sip.conf |
ASTERISK-03290: Dial returns CONGESTION instead of CHANUNAVIAL if host not reachable |
ASTERISK-03291: [patch] Misc grammar and spelling fixes |
ASTERISK-03292: [patch] Allow mailbox lookups into astdb (compile-time option) |
ASTERISK-03293: mpg123 Consuming 99% of the CPU |
ASTERISK-03294: [patch] allow to see who is talking in a conference |
ASTERISK-03295: Crash in transfer |
ASTERISK-03296: IAX doesn't honor the "bindaddr" directive |
ASTERISK-03297: [patch] resolve compiling warning |
ASTERISK-03298: [patch] app_dial Fix |
ASTERISK-03299: [patch] after #3353 fix h.323 channel doesn't build |
ASTERISK-03300: sip show peers output sometimes truncates first character on peer/user name |
ASTERISK-03301: Gastman-- CallerID Name no longer displayed |
ASTERISK-03302: Gastman command window history mechanism causes crashes, needs work |
ASTERISK-03303: [patch] Avoid cvs update warning about .depend |
ASTERISK-03304: [patch] Some more coding guidelines |
ASTERISK-03305: [feature] stayinmediapath=on/off command for SIP |
ASTERISK-03306: [patch] Configurable options for cdr_csv |
ASTERISK-03307: Using "send text" to a SIP phone crashes Asterisk |
ASTERISK-03308: incorrect loading of dialplan from database |
ASTERISK-03309: [patch] RetryDial fix |
ASTERISK-03310: Available variables not mentioned in sample config file |
ASTERISK-03311: No MOH when agents puts caller on hold. |
ASTERISK-03312: Voicemail Sample config has wrong default value for email Subject: |
ASTERISK-03313: [patch] Fix mpg123 build on non-Linux platforms |
ASTERISK-03314: [patch] Add to Makefile the ability to check pwlib/openh323 versions |
ASTERISK-03315: [patch] IAX2 codecpriority Update and bug fix |
ASTERISK-03316: cdr_addon_mysql badly assumes table will always be 'cdr' |
ASTERISK-03317: [patch] Add new driver to README |
ASTERISK-03318: [patch] makes init script portable for Debian |
ASTERISK-03319: ztdummy.c checks for LINUX26 before including zaptel.h |
ASTERISK-03320: [patch] Warn the user about the wrong mpg123 version |
ASTERISK-03321: [patch] flag to record a conference pseudo channel only |
ASTERISK-03322: Two simultaneous connections to a mailbox: only one survives |
ASTERISK-03323: Call Parking Troubles |
ASTERISK-03324: [patch] add playbackonly mode for half-duplex OSS |
ASTERISK-03325: Call Parking Troubles |
ASTERISK-03326: [patch] res_monitor fix |
ASTERISK-03327: [patch] Send timestamps with trunk frame entries |
ASTERISK-03328: Native assisted transfer atxfer looses language settings |
ASTERISK-03329: [patch] MeetMe enhancements |
ASTERISK-03330: re-open 3401: Native assisted transfer atxfer looses language settings |
ASTERISK-03331: [patch] Typo fix |
ASTERISK-03332: [patch] Outcalling from within meetme |
ASTERISK-03333: [PATCH] redesign config system interaction with res_config modules |
ASTERISK-03334: Crash on an unknown situation related to caller-id |
ASTERISK-03335: [patch] Advancec configuration file for meetme |
ASTERISK-03336: [patch] Remove channels/alaw.h from sources |
ASTERISK-03337: [patch] Asterisk crashes when loading chan_modem_i4l. |
ASTERISK-03338: Unresolved symbol in chan_modem_i4l.c stop asterisk (Not a bug, missing change in code) |
ASTERISK-03339: ISDN4Linux driven devices only present a lot of noise. |
ASTERISK-03340: inclusion of rpm spec file |
ASTERISK-03341: pound sign not recognized |
ASTERISK-03342: [patch] Small compatibility fixes for Q.931 implementation |
ASTERISK-03343: Pushing Transfer Button Breaks ASTCC Billing |
ASTERISK-03344: [patch] app_sipredirect for generating '302 Moved Temporarily' messages |
ASTERISK-03345: [request] Support callerid with realtime extensions |
ASTERISK-03346: Setting OUT_RDNIS in stable? |
ASTERISK-03347: PrivacyManager Doesn't Intercept Calls |
ASTERISK-03348: [patch] Remove unused variables in res_odbc.c |
ASTERISK-03349: [patch] registration stops on fail return from create_addr() (e.g. DNS error) |
ASTERISK-03350: Problem in $CVSROOT/asterisk/include/asterisk/config.h |
ASTERISK-03351: Problem in $CVSROOT/asterisk/include/asterisk/util.h |
ASTERISK-03352: [patch] Fix typo in show_codec_n in frame.c |
ASTERISK-03353: Audio Packets Being Sent Prior to IAX ANSWER |
ASTERISK-03354: chan_h323 fails to accept incoming calls |
ASTERISK-03355: Complete list of UK indication tones |
ASTERISK-03356: [patch] API for vendor-specified CDR fields |
ASTERISK-03357: Setting md5secret and removing secret from a SIP peer in sip.conf causes both to be set. |
ASTERISK-03358: [patch] - res_config_mysql updates |
ASTERISK-03359: [patch] Festival 1.95 patch for asterisk |
ASTERISK-03360: [patch] Readable description of expired Q931 timer |
ASTERISK-03361: insecure= does not function as anticipated |
ASTERISK-03362: DIALEDPEERNUMBER incorrectly set |
ASTERISK-03363: Atxfer doesn't work (with oh323) |
ASTERISK-03364: [patch] Return uniqueid with astman event OriginateSuccess |
ASTERISK-03365: [patch] CLI command to force agent logoff |
ASTERISK-03366: [request] Parameterized CDRs |
ASTERISK-03367: resctrictcid always true |
ASTERISK-03368: Problem with vpb channel driver extension handling |
ASTERISK-03369: [patch] stop app_dial from changing caller*id |
ASTERISK-03370: INFO dmtf processing doesn't handle A, B and C tones |
ASTERISK-03371: [patch] - RealTime Extensions gets segfault when using res_config_mysql |
ASTERISK-03372: [patch] More descriptive output based on cause code |
ASTERISK-03373: [patch] ROSE invoke DivertingLegInformation2 operation support |
ASTERISK-03374: [patch] let app_queue inherit MONITOR_EXEC and MONITOR_EXEC_ARGS from calling channel, if they exist. |
ASTERISK-03375: [branch] adds support for SERVice maintenance messages |
ASTERISK-03376: Asterisk crashes on divison by zero in calc_timestamp (chan_iax2) |
ASTERISK-03377: SayNumber() pronouncing numbers wrongly in portuguese |
ASTERISK-03378: New batch of sounds from Allison |
ASTERISK-03379: [patch] new events for queue activity |
ASTERISK-03380: [patch] manager actions 'dbget' and 'dbput' |
ASTERISK-03381: [patch] manager hold events |
ASTERISK-03382: [patch] Makefile in asterisk-sounds is slightly broken. |
ASTERISK-03383: res_perl no longer compiles (config_pvt.h) cvs head |
ASTERISK-03384: [patch] Add peers summary line (like chan_sip) and improve console logging |
ASTERISK-03385: [patch] New application - SetRDNIS |
ASTERISK-03386: [patch] banal error into indications.conf |
ASTERISK-03387: [patch] Added CLI tab completion for "iax2 show peer <peername>" |
ASTERISK-03388: [patch] Makefile update for Fedora AMD64 |
ASTERISK-03389: Config.c 1.53 crashes on start up |
ASTERISK-03390: The "g parameter" doesn't work |
ASTERISK-03391: [PATCH] astxs does not use ASTSRC for include paths |
ASTERISK-03392: Playing music on hold does not work - results in Floating point exception |
ASTERISK-03393: Cannot log Agent off |
ASTERISK-03394: A South African "tone zone" [za] |
ASTERISK-03395: sip conversation falls after 17 seconds... |
ASTERISK-03396: Manager interface outputs caller id of called, instead of caller one |
ASTERISK-03397: Regular Expression matching operator : not working as described in the documentation. |
ASTERISK-03398: SIP calls hangup with CVS from at least 21/01/05 onwards |
ASTERISK-03399: [patch] add summary line to 'dundi show peers' |
ASTERISK-03400: Application to get sip inuse counter from within dialplan |
ASTERISK-03401: 'System' missing |
ASTERISK-03402: Loud squealing noise heard on FXS hangup for the duration of flashtime |
ASTERISK-03403: [request] don't quelch dtmf when bridging between two inband methods. |
ASTERISK-03404: segmentaion fault in app_meetme.c |
ASTERISK-03405: Race condition in ast_request_and_dial |
ASTERISK-03406: [patch] proper cleanup on exit of last user from MeetMe |
ASTERISK-03407: Outgoing SIP unable to lookup SRV record host name |
ASTERISK-03408: [patch] Minor typo in sip history usage |
ASTERISK-03409: Realtime updating username feild to 's' |
ASTERISK-03410: [patch] #exec (an SSI like directive) |
ASTERISK-03411: non-numerical calleridnum can't be presented |
ASTERISK-03412: [design discussion] store/retrieve messages on/from IMAP server |
ASTERISK-03413: [patch] app_meetme fails to send audio |
ASTERISK-03414: [patch] Always asking user for a password in app_meetme |
ASTERISK-03415: include calleridname when app_dial changes callerid |
ASTERISK-03416: crash on transfer |
ASTERISK-03417: SIP digest auth replaces FROM: with what's in TO: |
ASTERISK-03418: [patch] implement prefixes for CLID on PRI channels based on type of number |
ASTERISK-03419: Missing Caller ID when using manager API the caller ID is missing |
ASTERISK-03420: Error message for execincludes inconsistent with config syntax |
ASTERISK-03421: [patch] Dependencies from .depend is valid from static library only, not for shared |
ASTERISK-03422: [PATCH] Bug in ast_write_file produces invalid WAV files when appending audio. |
ASTERISK-03423: [patch] Add last sent/received command to "sip show channels" |
ASTERISK-03424: [patch] fix incompatibility with phone switch (pridialplan=dynamic) |
ASTERISK-03425: [patch] Keeping CDR time in GMT |
ASTERISK-03426: Crash on iax call to empty queue (probable cause is moh) |
ASTERISK-03427: latest cvs head compile fails for openbsd |
ASTERISK-03428: [patch] make process fixes and .cvsignore update |
ASTERISK-03429: app_addon_sql_mysql loses connection to mysql often |
ASTERISK-03430: Minor annoyance in format_g729.c |
ASTERISK-03431: [patch] Segfault in IAX2 with music on hold |
ASTERISK-03432: [patch] SetCallerID bad assignation |
ASTERISK-03433: [patch] adds app_record 'b' option to stop beep tone |
ASTERISK-03434: [patch] IAX and SIP Realtime Improvement |
ASTERISK-03435: [patch] adds -s option to show uptime to display uptime seconds |
ASTERISK-03436: [patch] [post-1.4] updates on res_odbc |
ASTERISK-03437: no MOH to callback agents followed by blocking state |
ASTERISK-03438: Pressing keys during playback of ControlPlayback causes Asterisk to crash and burn. |
ASTERISK-03439: [patch] add cli 'show features' to dump feature map |
ASTERISK-03440: use snprintf instead of sprintf when filling variables..... |
ASTERISK-03441: [new app] app_readpipe |
ASTERISK-03442: cannot transfer calls picked up by callback agents |
ASTERISK-03443: Calling a SIP phone with no callerID at all segfaults |
ASTERISK-03444: [request] Add option to agents to NOT to do native transfer |
ASTERISK-03445: Forwarding unscreened CLIP and network provided CLIP to the dailplan |
ASTERISK-03446: Format_MP3 segfault fix for bug 3316 |
ASTERISK-03447: [patch] Send vars across a call |
ASTERISK-03448: [patch] Add debug level setting to asterisk.conf [options] |
ASTERISK-03449: [patch] README.variables update |
ASTERISK-03450: [patch] Update to README.cdr |
ASTERISK-03451: [patch] Configuration file README |
ASTERISK-03452: [patch] Sanity check in res_odbc |
ASTERISK-03453: agent acknowledgement not working |
ASTERISK-03454: [patch] Fix deadlock debugging |
ASTERISK-03455: [patch] Redirect/Transfer Issue |
ASTERISK-03456: when stop zaptel, produce a EIP message and don't unload modules |
ASTERISK-03457: [patch] Typos |
ASTERISK-03458: Asterisk CVS stable (current) crashes on remote user pressing # or * |
ASTERISK-03459: [patch] Typos in app_dial |
ASTERISK-03460: [patch] Update of README.configuration (multiline comments) |
ASTERISK-03461: Background() does not respond to touchtones A, B, C, or D |
ASTERISK-03462: missing subsystem id in wcte11xp.c ? |
ASTERISK-03463: mutex freed more times than locked |
ASTERISK-03464: [patch] Optimize search of builtin variables |
ASTERISK-03465: You cannot define mailboxes in an #include file |
ASTERISK-03466: app_queue does not pass stored Queue url to chan_agent for dial |
ASTERISK-03467: [patch] chap_zap.c in CVS doesn't compile on gcc 2.95.4 |
ASTERISK-03468: [patch] Update of README.configuration |
ASTERISK-03469: Asterisk deadlocks from time to time |
ASTERISK-03470: [patch] The ISDN Baerer Capability Information is not properly passed on a bridged channel |
ASTERISK-03471: [patch] Sanity checks and improvements |
ASTERISK-03472: Patch with BugID 3541 disables several Variables |
ASTERISK-03473: IVR API for apps |
ASTERISK-03474: [patch] Add Portugal to tone zone data |
ASTERISK-03475: [patch] IAX codec passthrough configuration |
ASTERISK-03476: [patch] libpri features - 2B channel transfer, MWI, callername over facility |
ASTERISK-03477: Typo fix in channel.c for "help show channeltypes" |
ASTERISK-03478: [patch] Added a new channel var to Monitor() |
ASTERISK-03479: [stable only] CallerID broken for me in latest chan_sip.c |
ASTERISK-03480: make /var/run/asterisk.ctl configurable |
ASTERISK-03481: [patch] wrong var used in app_dial m() option parser |
ASTERISK-03482: [patch] wrong var used in res_features config parser |
ASTERISK-03483: [patch] Simple ast_log filter |
ASTERISK-03484: [patch] http://www.digium.com/bugguidelines.html file is broken |
ASTERISK-03485: Bugfix 3519 introduced new bug for me: No CallerID number instead CID Asterisk |
ASTERISK-03486: dial_exec_full errors |
ASTERISK-03487: [patch] Remote MOH |
ASTERISK-03488: Segmentation fault upon invalid IAX peer registration |
ASTERISK-03489: [patch] adds note about crc-ccitt for 2.6 kernels |
ASTERISK-03490: Server stops processing IAX registrations after <?> hours of uptime |
ASTERISK-03491: [patch] Variables, accountcode for peers |
ASTERISK-03492: [patch] Add amaflags to peer, new cli "sip show user <name>" |
ASTERISK-03493: [patch] Fixes negative len in colon ${EXTEN:-4) |
ASTERISK-03494: [PATCH] rework channel structure to save more memory |
ASTERISK-03495: [PATCH] add completion to "sip show user" |
ASTERISK-03496: [PATCH] recent change to AST_DIGIT_ANY causes incorrect "user=phone" to be sent |
ASTERISK-03497: [patch] Fix chanvars at peer reload |
ASTERISK-03498: [patch] Add callingpress, incominglimit to peer |
ASTERISK-03499: [patch] Agent hangup during announcement is broken |
ASTERISK-03500: [patch] custom moh on CVS-Stable just looks for .mp3 files |
ASTERISK-03501: Looking up VM config from MySQL seems to be broken in stable Asterisk |
ASTERISK-03502: [patch] Improve call limit handling |
ASTERISK-03503: calls cut off when callers over iax leave messages |
ASTERISK-03504: [patch] Allow register within [peer] |
ASTERISK-03505: simple program based on zttool to display status of the cards |
ASTERISK-03506: [patch] No way for conferencee to know how many other in this conference |
ASTERISK-03507: [Patch] Add settable context option for outbound Dial() calls |
ASTERISK-03508: Asterisk Segmentation Fault - Upon rejected IAX peering request |
ASTERISK-03509: [patch] Estonian tonezone data |
ASTERISK-03510: Crash on unknown situation |
ASTERISK-03511: [h323 only] segfault and deadlock when rtp starts immediatelly |
ASTERISK-03512: atxfer lost digits entered while playing "pbx-transfer" |
ASTERISK-03513: Asterisk seg faults |
ASTERISK-03514: IAX2 bug when handling AddQueueMember, RemoveQueueMember |
ASTERISK-03515: [patch] Let CDR have variables |
ASTERISK-03516: [patch] distinctive ring detection with Caller ID/Australia. |
ASTERISK-03517: [patch] cdr_tds not tolerant of database server connectivity problems |
ASTERISK-03518: Asterisk Segfaults when using attached rules with chan_capi |
ASTERISK-03519: Audio delay in MeetMe using SIP when not 'q' mode |
ASTERISK-03520: [request] Presence IM client and connection to Live Communications Server 2005 |
ASTERISK-03521: When joining parties via Guided Transfer, the error below is reported. |
ASTERISK-03522: Asterisk changing 'From:' field after authorization challenge. |
ASTERISK-03523: [patch] Manager action 'DBGet' responds on all sessions |
ASTERISK-03524: [patch] app_md5: Generate md5 hash |
ASTERISK-03525: [patch] rework of retrieve_file() for ODBC VM Storage |
ASTERISK-03526: [request] Need to be able to cancel an Attended Transfer |
ASTERISK-03527: [patch] Fix the annoying misspelling of 'separate' |
ASTERISK-03528: Added SIPALERTINFO variable to contain the Alert-Info header valuer |
ASTERISK-03529: [patch - bugfix] Requests inside a dialog and strict routing vs. loose routing |
ASTERISK-03530: [patch] Monitor descriptions for manager actions |
ASTERISK-03531: [patch] Add event from agent logoff |
ASTERISK-03532: Disabling SIP-SIP RTP call bridging with reinvite |
ASTERISK-03533: Monitor command no longer accepts colons in filename |
ASTERISK-03534: [patch] - remove unused variables in GSM source |
ASTERISK-03535: zaptel.init fails regularly on FC3 (/dev/zap doesn't exist) |
ASTERISK-03536: Does asterisk respond correctly when it receives a 491 Request Pending |
ASTERISK-03537: [patch] channel driver UNISTIM |
ASTERISK-03538: CDRs not being written when DB is restarted. |
ASTERISK-03539: [patch] Add md5check for md5 verification |
ASTERISK-03540: [patch] add manager events for various parked call events |
ASTERISK-03541: Account code not populated when 'type=friend' sets caller*ID number |
ASTERISK-03542: Allow libpri to process Q931_IE_KEYPAD_FACILITY in state Q931_CALL_STATE_ACTIVE |
ASTERISK-03543: Allow libpri to process Q931_IE_KEYPAD_FACILITY in state Q931_CALL_STATE_ACTIVE |
ASTERISK-03544: SIP User permanently bound to IP |
ASTERISK-03545: [patch] app_voicemail fails to build in HEAD |
ASTERISK-03546: [PATCH] activate/deactivate dialplan contexts automatically |
ASTERISK-03547: [patch] Make Sure IAX2 picks the requested format (if available) |
ASTERISK-03548: cdr_pgsql does not gracefully handle database restarts |
ASTERISK-03549: Accountcode setting in sip.conf not being stored in CDRs for certain SIP clients. |
ASTERISK-03550: [patch] allows 0 second retry intervals |
ASTERISK-03551: [patch] Add "show hints" CLI command |
ASTERISK-03552: IAX registration request does not timeout properly |
ASTERISK-03553: New App ExecIfTime() |
ASTERISK-03554: Fax extension doesnt work with macros? |
ASTERISK-03555: false Hangup detected on bridged Zap calls |
ASTERISK-03556: [patch] $(func "arg" xyz) expression in dialplan |
ASTERISK-03557: Square brackets in pattern matching produces pbx_config : out of stack |
ASTERISK-03558: [patch] Incorrect parsing of calling party ID when it is withheld |
ASTERISK-03559: [patch] gcc 2.96 no longer compiles asterisk |
ASTERISK-03560: [patch] Add devicestate notification to IAX2 |
ASTERISK-03561: [patch] crash introduced in mysql_vm_routines.h |
ASTERISK-03562: [patch] ast_say_date_with_format problem in Spanish |
ASTERISK-03563: Asterisk blocks only h323 bound calls, all others look good |
ASTERISK-03564: [patch] Changes to devstate, message and subscribe |
ASTERISK-03565: [patch] Documentation for manager commands |
ASTERISK-03566: [patch] app_meetme admin pin patch breaks parsing of meetme.conf |
ASTERISK-03567: [PATCH] make member persistence future-proof |
ASTERISK-03568: [PATCH] add API for parsing/describing caller presentation values |
ASTERISK-03569: [patch] Default Qualify= setting in [general] of sip.conf |
ASTERISK-03570: Dial with g option causes asterisk to crash after # transfer |
ASTERISK-03571: [patch] Dialstring fix |
ASTERISK-03572: Agent groups not working |
ASTERISK-03573: [Patch] Allow -T option for asterisk startup to output everything timestamped. |
ASTERISK-03574: [patch] register with just a peer name use wrong REGISTER line |
ASTERISK-03575: [patch] Running out of RTP ports due to SUBSCRIBEs (No RTP ports Remaining) |
ASTERISK-03576: [patch] app_queue should reset timeout when wait_for_answer() finds busy |
ASTERISK-03577: [patch] queue_log AGENTCALLED logging |
ASTERISK-03578: [patch] Fix improper ignoring of second SIP INVITE messages on slow links |
ASTERISK-03579: [patch] Generic Events System |
ASTERISK-03580: [PATCH] don't do codec matching until we know who the caller is |
ASTERISK-03581: [patch] Add the total users into a conf. |
ASTERISK-03582: [patch] allow asterisk to compile on gcc4 |
ASTERISK-03583: [patch - bugfix] False REGISTER denials with unreliable transport. |
ASTERISK-03584: Realtime updates |
ASTERISK-03585: [patch] README.realtime |
ASTERISK-03586: [patch] Pluggable PBX's |
ASTERISK-03587: [patch] Mexican tonezone data |
ASTERISK-03588: SetVar directive not functioning in callfiles |
ASTERISK-03589: [request] Brazilian Caller ID Detection - DTMF without polarity reversal |
ASTERISK-03590: app_system additions. |
ASTERISK-03591: in do_monitor a call to ast_sched_runq(sched) never returns |
ASTERISK-03592: [Patch] Maximum users for MeetMe |
ASTERISK-03593: Seg fault when connecting to Asterisk console |
ASTERISK-03594: MGCP CDR transfer bad |
ASTERISK-03595: [patch] rfc2833 DTMFs sent with bad timestamps |
ASTERISK-03596: Call acknowledgement for a Directed Dial to a specific Agent. |
ASTERISK-03597: [patch] AGI Status in exec'ed cmds |
ASTERISK-03598: [patch] show extension <exten> |
ASTERISK-03599: realtime authentication failure with chan_sip with type != friend |
ASTERISK-03600: [patch] TXTCIDName() lookup mangles response |
ASTERISK-03601: Goto and GotoIf do not work with a RealTime extension list |
ASTERISK-03602: [patch] app.c will not compile |
ASTERISK-03603: [patch] adds alias for wctdm, and adds keys for te110p |
ASTERISK-03604: Addition of autousupport script into stable |
ASTERISK-03605: [patch] Voicemail crash on vm_authenticate() |
ASTERISK-03606: [patch] Small change in "show manager commands" for agent command |
ASTERISK-03607: Ability to forward voicemail to multiple recipients with the same prepend message. |
ASTERISK-03608: [patch] Doxygen docs for manager.h |
ASTERISK-03609: chan_phone phone_send_textreports wrong error code |
ASTERISK-03610: [patch] New manager commands for SIP (Not XML any more) |
ASTERISK-03611: [patch] Fixed manager IAXpeers, add IAXshowpeer in manager format |
ASTERISK-03612: Ringback doesn't work for SIP or IAX channels. |
ASTERISK-03613: smsq in PortugalTelecom |
ASTERISK-03614: [patch] Config sample updates - reloads |
ASTERISK-03615: [patch] ENUm and DNS docs |
ASTERISK-03616: [patch] Updates to Mandrake init scripts |
ASTERISK-03617: MOH stopped working properly in CVS Head |
ASTERISK-03618: App_queue/chan_agent blocks after redirect |
ASTERISK-03619: Add StartMusicOnHold and StopMusicOnHold |
ASTERISK-03620: [patch] find out if udev or devfs is running for good |
ASTERISK-03621: [patch] get more useful info from zttest |
ASTERISK-03622: [patch] utils.c does not compile |
ASTERISK-03623: [patch] adds ability to limit more manager events |
ASTERISK-03624: [patch] allows any DTMF to disconnect an agent, not just * |
ASTERISK-03625: [patch] H.263+ video codec support (passthru) |
ASTERISK-03626: [patch] Consultative transfers between asterisk servers |
ASTERISK-03627: Codec mismatch problem |
ASTERISK-03628: no difference between strategy roundrobin & rrmemory |
ASTERISK-03629: RTP stack doesn't see incoming packets unless outgoing packet is sent |
ASTERISK-03630: [patch] pridialplan=dynamic fix prevents * startup when no libpri installed |
ASTERISK-03631: MOH to remote is dropped after parking call |
ASTERISK-03632: DeadLock related to Zap (seems) |
ASTERISK-03633: [patch] Can't send text from extensions shared with "switch" before answered |
ASTERISK-03634: Ability to bridge a call from the Manager API |
ASTERISK-03635: fewer extra debugging information under 2.6.x |
ASTERISK-03636: Reverse Battery Should Not Equal Disconnect |
ASTERISK-03637: breaks 401 invite authentication |
ASTERISK-03638: DNS Error prevents SIP module from functioning correctly. |
ASTERISK-03639: [patch] utils.c will not compile |
ASTERISK-03640: AGI Command GET VARIABLE Delays output |
ASTERISK-03641: chan_sip does not support pre-authenticated re-REGISTER/does not deal with non-compliant gateway |
ASTERISK-03642: [request] Full support of SRV records |
ASTERISK-03643: buggy phone caused * to crash |
ASTERISK-03644: [patch] Cancel an in-progress attended transfer. |
ASTERISK-03645: Multiple SIP Registrations not being handled properly |
ASTERISK-03646: [patch] New variable FIELDQTY |
ASTERISK-03647: extensions.conf: linked list priorities |
ASTERISK-03648: Contact directory for voicemail |
ASTERISK-03649: Add Message Waiting Indicator arguments to mailbox options |
ASTERISK-03650: [request] Compose/Forward to AMIS subscriber |
ASTERISK-03651: Simple voicemail delivery |
ASTERISK-03652: Unable to compile chan_h323 |
ASTERISK-03653: agi running on two channels second channel gets interrupted system call when first channel hangs up |
ASTERISK-03654: [patch] JabberPops |
ASTERISK-03655: [patch] Makefile change to allow compiling using gcc4 |
ASTERISK-03656: ztcfg -vvv has problem when all channels are dacs or dacsrbs |
ASTERISK-03657: Calls coming from a queue can't be recorded using Monitor |
ASTERISK-03658: 1.0.7 Release Candidate - Please test and report! |
ASTERISK-03659: [patch] Should be able to turn off queue_log |
ASTERISK-03660: Asterisk randomly crashs |
ASTERISK-03661: [patch] Allow multiple files for the Playback app |
ASTERISK-03662: 4th Invite Causes One Way Voice or No Voice |
ASTERISK-03663: [patch] When a SIP client unregisters with asterisk, call states aren't properly handled using app_realtime |
ASTERISK-03664: [patch] Compilation problem with cdr_sqlite |
ASTERISK-03665: When use Asterisk manager to originate calls, channel.c send a warning and don't make the call |
ASTERISK-03666: leavewhenempty and joinempty do not understand when no agents are logged in |
ASTERISK-03667: Install the Asterisk 1.0.7 error |
ASTERISK-03668: The SIP terminal can not register asterisk server |
ASTERISK-03669: [patch] check for user agent before allowing authentication |
ASTERISK-03670: Rev 1.416 of chan_zap.c breaks compile |
ASTERISK-03671: Restore read format on exit in WaitForSilence |
ASTERISK-03672: [patch] Fix Makefile to compile codec_speex against system-install of 1.1.x |
ASTERISK-03673: asterisk.8 should not be compressed |
ASTERISK-03674: Asterisk security problem: authorized SIP users can fake any callerid! |
ASTERISK-03675: [PATCH] adds support for applications starting as feature |
ASTERISK-03676: Add inband audible ringback option to Dial |
ASTERISK-03677: "help show config mappings" fails |
ASTERISK-03678: IAX2 Reload |
ASTERISK-03679: [patch] say date support for AGI |
ASTERISK-03680: userfield not set in Local channel |
ASTERISK-03681: MGCP *70 disable call-waiting does not reactivate on endpoint hangup |
ASTERISK-03682: MGCP *69 does not call back correct number |
ASTERISK-03683: chan_zap does not wait for wink on E&M Wink lines |
ASTERISK-03684: zaptel.ko causes kernel panic when unloading |
ASTERISK-03685: Fixed Read in * 1.0.6 |
ASTERISK-03686: [patch] add newline to NOTICE |
ASTERISK-03687: IAX peers with more that 15 characters in name |
ASTERISK-03688: Unable to Re-Park calls that have timed out |
ASTERISK-03689: Incorrect SIP Channel |
ASTERISK-03690: [patch] Builtin ($if) function |
ASTERISK-03691: [patch] Standard UUID/GUIDs for CDR uniqueid |
ASTERISK-03692: [patch] Reorganize source code for handle_request |
ASTERISK-03693: [patch] Add comments on hints in pbx.c |
ASTERISK-03694: [patch] Beginning of voice prompts chopped off |
ASTERISK-03695: [patch] SIP debug output improvements |
ASTERISK-03696: [patch] new option G(context^exten^priority) |
ASTERISK-03697: Realtime Application : Probleme and error message. |
ASTERISK-03698: outboundproxy not working |
ASTERISK-03699: [patch] Small corrections to sipredirect 302 |
ASTERISK-03700: [patch/STABLE] Complain, but don't explode, if remote CVS-HEAD sends trunk timestamps |
ASTERISK-03701: [patch] Change the matching for incoming calls |
ASTERISK-03702: [patch] SIP inuse bug |
ASTERISK-03703: [patch] Don't destroy call until all responses on BYE have been processed |
ASTERISK-03704: AGI scripts defunct after exit (in FC3 only) |
ASTERISK-03705: [patch] INVITE to Polycom 500 broken after handle_request patch |
ASTERISK-03706: When call is destroyed, 487 "Request Cancelled" packets seem to cycle around for too long |
ASTERISK-03707: [patch] deadlock in app_queue compare_weight() |
ASTERISK-03708: Polycom 600 hangup generates Internal Server Error from phone |
ASTERISK-03709: [patch] SUBSCRIBE causes "Internal Server Error" on Polycom |
ASTERISK-03710: Don't send 100 Trying during registration transaction |
ASTERISK-03711: [patch] Don't send tag in 100 Trying responses |
ASTERISK-03712: [patch] SIP methods MUST be parsed case sensitive |
ASTERISK-03713: Cannot call ForkCDR more than once |
ASTERISK-03714: [patch] Channel variables not set on cloned channels during redirect |
ASTERISK-03715: [patch] More complete fix to address digit timeout on E&M T1 then emdigitwait |
ASTERISK-03716: [patch] RECAP OF #0003799 SUBSCRIBE causes "Internal Server Error" on Polycom |
ASTERISK-03717: [patch] chan_zap PRI progress handling optimization |
ASTERISK-03718: [request]: PBX identification for handle multiple PBX with one CDR DB |
ASTERISK-03719: [patch] Monitoring stops when channel redirected |
ASTERISK-03720: [patch] bindport statement seems not to function |
ASTERISK-03721: chan_sip not closing channel when RTP goes idle due to a faulty SIP device. |
ASTERISK-03722: [patch] README for jitterbuffer |
ASTERISK-03723: Busydetect failure on unanswered calls |
ASTERISK-03724: Segmentation Fault with Realtime |
ASTERISK-03725: [patch] meetme users unable to wait for marked user, no MOH for admin |
ASTERISK-03726: [patch] revert MOH part of #3815 |
ASTERISK-03727: Asterisk doesn't seem to cope well in dealing with retransmitted INVITES |
ASTERISK-03728: [patch 1.0] SIP accepts re-INVITE for T.38, thus terminating call |
ASTERISK-03729: [request] xfersound = beep for SIP transfers |
ASTERISK-03730: SIP INVITE Fail (407) after upgrading from 1.0.0 to 1.0.7 |
ASTERISK-03731: [PATCH] Allow user to exit out of the position announcement with a keypress |
ASTERISK-03732: [patch] Fix missing hangups in gastman |
ASTERISK-03733: Asterisk SIGPFE on inbound IAX call |
ASTERISK-03734: OpenBSD fails compile |
ASTERISK-03735: [patch] Driver 'Phone' does not have a fixup routine |
ASTERISK-03736: Call recording issue with new jitterbuffer |
ASTERISK-03737: [patch] Implement fax detection within i4l |
ASTERISK-03738: [patch] make rpm fails |
ASTERISK-03739: [patch] Detailed dialplan result from Queue() application |
ASTERISK-03740: The opaque has returned! |
ASTERISK-03741: SIGCHLD from exiting AGI script hangs up other call |
ASTERISK-03742: [patch] Split huge language-specific stuff from say.c into different loadable modules |
ASTERISK-03743: ADSI breaks voicemail |
ASTERISK-03744: data for Poland in zonedata.c seems invalid |
ASTERISK-03745: core dump following cb_extensionstate action? |
ASTERISK-03746: [patch] Return of ChanSpy |
ASTERISK-03747: [patch] error playing wav49 format with Windows Media Player |
ASTERISK-03748: [patch] Formatting fixes, doc updates |
ASTERISK-03749: [patch] OSS and ALSA channel drivers use wrong endianness. |
ASTERISK-03750: In IAX, Dial's D() modifier sends only the first tone |
ASTERISK-03751: [patch] Attended call transfer doesn't get back (on busy or noanswer) |
ASTERISK-03752: [patch] Use ast_strcasestr instead of strcasestr |
ASTERISK-03753: [patch] rudimentary support for AOC (Advice of Charge) |
ASTERISK-03754: G726-16, G726-24, and G726-40 passthrough |
ASTERISK-03755: [patch] set register timeout at reload |
ASTERISK-03756: [patch] Find endian.h |
ASTERISK-03757: [patch] Inband DTMF Signalling Behavior (In 1.0.x and Head) |
ASTERISK-03758: 'Avoided initial deadlock' message and Asterisk stops to handle calls |
ASTERISK-03759: [patch] Do not increment rtp->seqno before debug output |
ASTERISK-03760: [patch] Improve REGISTER= handling |
ASTERISK-03761: chanspy crash asterisk |
ASTERISK-03762: chan_h323 locking active channels |
ASTERISK-03763: [patch] Reformat source code |
ASTERISK-03764: [patch] Generic rtp jitterbuffer for Asterisk |
ASTERISK-03765: Crash with FORKCDR & CVS 03/25/05 |
ASTERISK-03766: asterisk-addons and 64bit make |
ASTERISK-03767: [patch] ${DATETIME} format and documentation do not agree |
ASTERISK-03768: [patch] Restore old Caller ID behavior as the deafult |
ASTERISK-03769: DISA uses wrong timeout when no-password is used |
ASTERISK-03770: highpriority does not work in asterisk.conf |
ASTERISK-03771: Agent & queue locks if using local channels |
ASTERISK-03772: [patch] AGIs continue even if a file was not found with GET OPTION |
ASTERISK-03773: [patch] swap location of 2 config options to place them under their paragraph |
ASTERISK-03774: [patch] ExecIfTime Always returns zero |
ASTERISK-03775: [patch] AST_FORMAT_SLINEAR has ambigious endianness. |
ASTERISK-03776: [patch] Sending Polarity/DTMF Caller ID in chan_zap |
ASTERISK-03777: [patch] Architectures without unaligned accesses unable to properly negotiate codecs |
ASTERISK-03778: [patch] Asterisk support for cross compilation |
ASTERISK-03779: [patch] use of parsing api for app_skel and more |
ASTERISK-03780: [patch] simple load testing app |
ASTERISK-03781: Using a precalculated tone is easier on the processor than calculating it mathmatically |
ASTERISK-03782: [patch] Create new VM_CONTEXT for subsitution variables |
ASTERISK-03783: ChanISAvail not working properly |
ASTERISK-03784: [Patch] [post 1.2] answer and hungup on polarity switch Fix (works in SPAIN) |
ASTERISK-03785: [patch] compile failure with (development) gcc 4 |
ASTERISK-03786: [patch] iax2 reload, fix counters for iax2 show peers |
ASTERISK-03787: "sip notify" for Cisco SIP needs extra newline in NOTIFY message body |
ASTERISK-03788: [patch] AGIs continue even if a file was not found with GET DATA |
ASTERISK-03789: [patch] enter/leave prompt easily changable.. Again ;) |
ASTERISK-03790: [patch] meetme silence supression (using Monitortalker) |
ASTERISK-03791: [patch] Add E option to the Dial command to explicitely disable echo cancellation |
ASTERISK-03792: [patch] implement sending of 'early'-state in NOTIFY-Messages |
ASTERISK-03793: [patch] detach CDRs for posting in a separate thread |
ASTERISK-03794: [patch] little fix with SAY NUMBER |
ASTERISK-03795: find_call fails (pedantic=yes): improper tag checking on outgoing call |
ASTERISK-03796: [patch] ztdummy support in zaptel.init |
ASTERISK-03797: [patch] syslog facility in logger.c is not chosen correctly |
ASTERISK-03798: using "c" to count users does not play entry beep |
ASTERISK-03799: [patch] Dialplan lacks instant match for unambiguous wildcard. |
ASTERISK-03800: [patch] logger.conf queue_log doesn't work |
ASTERISK-03801: Incorrect "user information layer 1" representation |
ASTERISK-03802: Linux kernel crashes after insmod wct4xxp and ztcfg |
ASTERISK-03803: [new_app] app_dictate |
ASTERISK-03804: [patch] make format_sln default to write .raw files |
ASTERISK-03805: [patch] Makefile CC |
ASTERISK-03806: [patch] Selecting 'Do not ask again' should take the default in that dialog |
ASTERISK-03807: Mute status from CLI |
ASTERISK-03808: [request] Coach <->Pupil |
ASTERISK-03809: [patch] Caller Unable to mute themselves in meetme room |
ASTERISK-03810: [patch] Startup script for Slackware |
ASTERISK-03811: [patch] Buggy zaptel optimization patch |
ASTERISK-03812: [patch] Recent changes to app_queue will not compile with gcc-2.95 |
ASTERISK-03813: cdr_custom fails to load and stops Asterisk starting |
ASTERISK-03814: [patch] asterisk-addons mysql app fix |
ASTERISK-03815: [patch] Fix Makefile in asterisk-addons for "gmake" |
ASTERISK-03816: [patch] New CLI command - show cdrdrivers |
ASTERISK-03817: Asterisk crashes after find_callno() failure |
ASTERISK-03818: [patch] wrong usage of rtpchecksums |
ASTERISK-03819: [patch] Logger handling of no space on hard disk |
ASTERISK-03820: [patch] dtmf rfc2833 and senddigit |
ASTERISK-03821: [patch] Parking - Allow user to assign a macro when the call times out |
ASTERISK-03822: cdr_custom.so |
ASTERISK-03823: queue join_empty has wrong logic for 'strict' and 'yes' settings |
ASTERISK-03824: '#" not recognised in get_data (AGI mode) |
ASTERISK-03825: [patch] Compiling libpri with GCC 4 |
ASTERISK-03826: [patch] Improve "logger show channels" |
ASTERISK-03827: [patch] Add category to events |
ASTERISK-03828: realtime voicemail options from database not paid attention to |
ASTERISK-03829: [patch] pbx_spool |
ASTERISK-03830: Incomplete information in SIP Contact header |
ASTERISK-03831: Asterisk fails to properly authenticate the call from multi-fxs gateway |
ASTERISK-03832: originate manager command bug |
ASTERISK-03833: [patch] chan_sip does not decode escaped characters as per the RFC |
ASTERISK-03834: [patch] PBX.c formatting, typos |
ASTERISK-03835: [patch] app_voicemail - say caller name before listening to voicemail |
ASTERISK-03836: [patch] Add Asterisk variable that contains file name and path of new voicemail |
ASTERISK-03837: [patch] pbx_spool feof vs fgets EOF detection |
ASTERISK-03838: [patch] Don't hold channel list lock when it's not needed |
ASTERISK-03839: [patch] Remove coloring escape sequences from file/syslog logs |
ASTERISK-03840: [branch][post 1.4] add manager "Listdialplan" command |
ASTERISK-03841: [patch] fix all deadlocks in app_queue and chan_agent. |
ASTERISK-03842: [patch] split meetme into more functions |
ASTERISK-03843: [patch] Improve REGISTER |
ASTERISK-03844: [patch] Asteriskify the formatting of jitterbuf.c |
ASTERISK-03845: [patch] Document manager headers in manager.h |
ASTERISK-03846: [patch] Compile warning for chmod |
ASTERISK-03847: Compile warnings |
ASTERISK-03848: Crash on start. |
ASTERISK-03849: *67 caller ID blocking on MGCP only works for first call |
ASTERISK-03850: Registration fails when IP address of peer changes |
ASTERISK-03851: zaptel.conf span= ignores first number |
ASTERISK-03852: asterisk compilation fails on chan_h323 |
ASTERISK-03853: Supervised transfer problems |
ASTERISK-03854: Register hangs using SRV records when host is down |
ASTERISK-03855: [patch] ast_waitstream does not close the stream. |
ASTERISK-03856: [patch] ast_say_digit_str skips minus |
ASTERISK-03857: [patch] ast_say_date_with_format_fr is incorrect for year |
ASTERISK-03858: [patch] adds hidefromdir option to app_directory |
ASTERISK-03859: [patch] MGCP "accountcode" option is not propagated in CDR |
ASTERISK-03860: Dead code in chan_iax2.c |
ASTERISK-03861: chan_iax2 reports wrong address and port. |
ASTERISK-03862: error in zaptel.init block module loading (Debian) |
ASTERISK-03863: Missing cmd chkconfig on Debian blocks make config |
ASTERISK-03864: ChanSpy is able to listen upon itself |
ASTERISK-03865: [request] "iax2 show registry" doesn't show prior success |
ASTERISK-03866: SIP login problems in "sip show registry" |
ASTERISK-03867: chan_iax2 not RFC compliant :p |
ASTERISK-03868: non-codec capabilities is printed in verbose incorrectly. 0x1 (g723) it should be telephone-event. |
ASTERISK-03869: iax2 trunking ping timestamp problem |
ASTERISK-03870: [patch] mgcp amaflags are not passed to cdr |
ASTERISK-03871: ALERTING not being sent for inbound PRI calls |
ASTERISK-03872: [patch] No joinwhenempty in app_queue |
ASTERISK-03873: [patch] fix to timestamps in jitterbuf and chan_iax2 |
ASTERISK-03874: sip show inuse has stopped working in latest cvs |
ASTERISK-03875: [patch] H.323 driver improvements |
ASTERISK-03876: cdr_custom is broken on head, and won't register. |
ASTERISK-03877: Data Calls through asterisk don't work |
ASTERISK-03878: Incorrect handling of: 183 Session in progress. No early media. |
ASTERISK-03879: [patch] MGCP *70 callwaiting disable does not reenable on hu |
ASTERISK-03880: Problems with DTMF not being received correctly |
ASTERISK-03881: [request] adding forwarded time and user to forwarded messages |
ASTERISK-03882: [patch] SIP RTP redirect problem after MoH |
ASTERISK-03883: [patch] voicemail saves wav49 format with wrong extension |
ASTERISK-03884: No MOH when agents presses Snom Hold button. |
ASTERISK-03885: Reload causes asterisk to die |
ASTERISK-03886: queues lack distinctive ring support |
ASTERISK-03887: [patch] joinempty options backwards |
ASTERISK-03888: H.323 codec negotiation fails |
ASTERISK-03889: [patch] Rework of make subsystem for chan_h323 to be compatible with OpenH323 |
ASTERISK-03890: [patch] Allow app_curl to be compiled with outdated libcurl |
ASTERISK-03891: memory leak in app_dial option for musiconhold ? |
ASTERISK-03892: [patch] say.c attempts to play files from the wrong location |
ASTERISK-03893: [patch] SAY DATETIME (SayUnixtime) support for AGI |
ASTERISK-03894: RTP port allocation fails |
ASTERISK-03895: on multihomed PC Asterisk bad response blocks calls |
ASTERISK-03896: [request] Set RDNIS from Diversion Header on 300 Messages |
ASTERISK-03897: [pending stable] voicemail saves wav49 format with wrong extension |
ASTERISK-03898: [pending stable] Data Calls through asterisk don't work |
ASTERISK-03899: [patch] Don't abort reconfiguration of other channels if we encounter a connected HDLC device |
ASTERISK-03900: Coredumps on outgoing H.323 slow start calls |
ASTERISK-03901: [patch] app_queue compilation broken on gcc 2.95 |
ASTERISK-03902: [patch] Send DTMF to calling party |
ASTERISK-03903: [patch] Add Member & Queue filters to QueueStatus manager command |
ASTERISK-03904: [patch] cdr_tds is not compatible with ver 0.63 of FreeTDS |
ASTERISK-03905: [request] make install rewrite /dev/zap/* to unusefull values |
ASTERISK-03906: [patch] WaitExten option for Music on Hold |
ASTERISK-03907: make does not compile libtonezone |
ASTERISK-03908: Content-Type header in "OK" response confuses asterisk |
ASTERISK-03909: make install crash for nonroot user |
ASTERISK-03910: Asterisk needs to be able re-initialise the RTP stream processing after a re-invite |
ASTERISK-03911: [request] make samples overwrites all config files without promt |
ASTERISK-03912: [request] support authenticated INFO |
ASTERISK-03913: Mailbox cannot start with letters "sbu" |
ASTERISK-03914: [patch] clean up some verbose to be correct. |
ASTERISK-03915: [patch] Causes.h formatting |
ASTERISK-03916: Multihomed PC Asterisk still does not working on SIP |
ASTERISK-03917: Call Parking times out to the wrong extension |
ASTERISK-03918: Music On Hold with Re-Invite |
ASTERISK-03919: Call pickup with SIP not working |
ASTERISK-03920: D-channel knocked down by remote zap channels |
ASTERISK-03921: asterisk-oh323 compile errors on Fedora Core Linux 3 |
ASTERISK-03922: [patch] Documentation for manager "redirect" command |
ASTERISK-03923: [patch] Update manager.txt with cdr_manager headers |
ASTERISK-03924: Callid generation between restarts in chan_sip |
ASTERISK-03925: [patch] Another newline beautification patch |
ASTERISK-03926: chan_alsa can't unload |
ASTERISK-03927: Registrations not tolerant of time changes |
ASTERISK-03928: crashes h.323/GK Nortel CS2K Carrier Softswitch |
ASTERISK-03929: add ast_sched_when() function |
ASTERISK-03930: [patch] count parked calls |
ASTERISK-03931: [patch] fix peer matching for multiple peers at the same IP address in 'insecure' mode |
ASTERISK-03932: Crash observed |
ASTERISK-03933: [patch] tone zone for India |
ASTERISK-03934: a couple missing \n's and some very minor misspellings |
ASTERISK-03935: [patch] Change parking operation to remember offset |
ASTERISK-03936: [patch] FastAGI will go at priority N+101 |
ASTERISK-03937: empty name in agents.conf causes rubbish in 'show agents' |
ASTERISK-03938: When specifying T or t in Dial cmd, Asterisk still attempts native bridge |
ASTERISK-03939: [patch] No hangup in exten h |
ASTERISK-03940: [patch] Conflict ztdummy with hardware modules |
ASTERISK-03941: [patch] Make jump to n+101 optional on Dial command |
ASTERISK-03942: Pattern matching with "ex-girlfriend" logic does not accept variable substitution |
ASTERISK-03943: [patch] realtime support for app_queue.c |
ASTERISK-03944: [patch] Allow incoming DIDs to be deflected to the "i" extension |
ASTERISK-03945: [patch] Build/compile fixes for zaptel |
ASTERISK-03946: application PGSQL() fails to build due to conflicting types |
ASTERISK-03947: [patch] ast_outgoing_* retains lock on sync > 1 |
ASTERISK-03948: [patch] extension of MeetMe's X option |
ASTERISK-03949: [patch]allow prev 4, next 6 to wrap when msg come to end |
ASTERISK-03950: International sounds support is broken |
ASTERISK-03951: 'Avoiding initial deadlock' message and Asterisk stops to operate |
ASTERISK-03952: [request] Outbound Calling Name by FACILITY IE |
ASTERISK-03953: core dump in libpri |
ASTERISK-03954: [patch] Allow transfers to a different server |
ASTERISK-03955: Create (hopefully) less confusing "Removing Characters from a String" docs |
ASTERISK-03956: Update CLI docs for DigitTimeout() and ResponseTimeout() indicating default timeout |
ASTERISK-03957: [PATCH] add __attribute__ ((packed)) in q921_xxx structure definitions |
ASTERISK-03958: deadlocks when manager connection dies without sending disconnect |
ASTERISK-03959: [patch] allow zaptel compile against any installed kernel, not just the currently running one |
ASTERISK-03960: subscriptions do not work any more |
ASTERISK-03961: [patch] goto_on_transfer for blind #transfer |
ASTERISK-03962: [patch] Add support for using labels in SET PRIORITY command |
ASTERISK-03963: [patch] always use #include "" for non-system headers |
ASTERISK-03964: [patch] Wait failed (random message) in ast_waitfordigit_full() |
ASTERISK-03965: [patch] Hang during call to Wait - also cause of deadlock messages |
ASTERISK-03966: make install error: .depend: No such file or directory |
ASTERISK-03967: asterisk and ethereal together crashes box |
ASTERISK-03968: IAX2 REJECT does not pass cause code back |
ASTERISK-03969: No DTMF detection from PSTN endpoint on outbound calls |
ASTERISK-03970: [patch] realtime update cli returns incorrect result |
ASTERISK-03971: [patch] put system headers before asterisk headers (part 1) |
ASTERISK-03972: [request] individual channel debug ability |
ASTERISK-03973: [patch] Display Names in SIP To: headers |
ASTERISK-03974: [patch] Add DND manager notification to chan_zap |
ASTERISK-03975: [patch] Changes of %i for %d in res_agi.c |
ASTERISK-03976: [patch] Fixes synopsis for some applications. |
ASTERISK-03977: [request] HANGUPCAUSE doesn't populate in chan_sip |
ASTERISK-03978: [patch] rtcache textual clearup attempt |
ASTERISK-03979: sip show peer load doesn't really load from ARA |
ASTERISK-03980: [Patch] chan_misdn patch for asterisk-head |
ASTERISK-03981: ResponseTimeout called from macro using Dial Application crashes Asterisk |
ASTERISK-03982: CDR fix for ASTCC |
ASTERISK-03983: single options column not supported in app_voicemail when using realtime |
ASTERISK-03984: [request] expand functionality of 'mailbox' option in sip.conf |
ASTERISK-03985: SetGroup persists after call is transfered |
ASTERISK-03986: Zap option 'c' breaks Dial option 't' |
ASTERISK-03987: Changing socket and .pid directory requires editing two files and recompiling |
ASTERISK-03988: [patch] New ringing-associated time boundaries for Dial |
ASTERISK-03989: [Patch] make update support for Asterisk-addons Makefile |
ASTERISK-03990: cdr_addon_mysql fails to build |
ASTERISK-03991: [patch] Small formatting fix for chan_sip |
ASTERISK-03992: [patch] iax, log remote ip on REGREJ |
ASTERISK-03993: Global variables do not get set when called from a macro by Dial application |
ASTERISK-03994: Zap option 'c' breaks Dial option 't' |
ASTERISK-03995: ResponseTimeout called from macro using Dial Application crashes Asterisk |
ASTERISK-03996: [patch] Add a superclean function to the Makefile |
ASTERISK-03997: Pattern matching with "ex-girlfriend" logic does not accept variable substitution |
ASTERISK-03998: [patch] Add a superclean function to the Makefile |
ASTERISK-03999: [patch] CDR_ODBC doesnt close the connection afther recording CDR. |